eac3to - audio conversion tool

[jpsdr]

Do you intend by any chance to update some " old DLL" (ffmpeg, flac, ...) to newer versions ?
For exemple, dcadec is now implemented into ffmpeg, so all improvements of it are not anymore in the standalone version.

[wiggaz]

I still get a corrupt video (Passengers) with the latest version (3.34). Am I missing something?
Thanks in advance.

[madshi]

I still get a corrupt video (Passengers) with the latest version (3.34). Am I missing something?
Can you post a small video extract with which I could reproduce the problem? Thanks!

[tormento]

Sometimes I get Invalid bitstream format error with DTS to AC3 conversion and I have to use ffmpeg.

I want to supply you an example (https://ufile.io/ifxhm), so probably a fix will be out.

eac3to.exe eng.dts eng.ac3
DTS Master Audio, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 1509kbps, 48kHz)
Decoding with libDcaDec DTS Decoder...
Remapping channels...
Encoding AC3 <640kbps> with libAften...
Creating file "eng.ac3"...
libDcaDec reported the error "Invalid bitstream format".
Aborted at file position 372244480.

[wiggaz]

Can you post a small video extract with which I could reproduce the problem? Thanks!
First of all, thanks for your answer.
Since I thought maybe it was the source corrupted, I tried muxing the m2ts directly and it was flawless, so it couldn't be the disc.
And that's why there are two sample: one muxing the .h265 extract with eac3to [corrupted], and the second muxing directly the m2ts (only the video of course) [flawless].
They are 120s each.

http://www.mediafire.com/file/ts3fsbjgzmrwwzc/eac3to_sample-001.mkv
http://www.mediafire.com/file/6x3tlidcloo1b30/mkvtoolnix_sample-001.mkv


eac3to is update to 3.34 with Useac3to at 1.28 and mkvtoolnix is update to 18.0.0.
Thanks in advance again.

[madshi]

Thanks. But what I really need is to be able to reproduce the problem on my PC, so I can double check if the problem is really fixed. So I need an extract of the m2ts file.

[wiggaz]

Thanks. But what I really need is to be able to reproduce the problem on my PC, so I can double check if the problem is really fixed. So I need an extract of the m2ts file.

The second file is the m2ts file muxed into mkv, with all the tracks disabled except of the video.
Or should I give you the mux with all the tracks, divided into a small piece (120sec like the others) with mkvtoolnix?

[madshi]

I don't need the mux, I need the original m2ts file (a small part of it)!

[sneaker_ger]

@wiggaz
You can use DGSplit (http://rationalqm.us/dgsplit/dgsplit12.zip) to cut e.g. 50 MB from the beginning of the m2ts.

[madshi]

That should work, but only if the corruption is right at the start of the movie?

[wiggaz]

Here's, splitted with DGSplit as suggested (thx sneaker_ger): http://www.mediafire.com/file/24om27uq5bukho3/00001_0.m2ts

And this is an example: http://thumbs2.imagebam.com/27/45/00/0beea0667232263.jpg (http://www.imagebam.com/image/0beea0667232263)

[madshi]

Thanks! And the problem also occurs with this small m2ts file, correct? (Don't have time to test right now.)

[sneaker_ger]

@wiggaz
I can't reproduce the problem with the "00001_0.m2ts". Only with "eac3to_sample-001.mkv". Please check again that you really are on eac3to 3.34 and not 3.32.

[wiggaz]

allright. I'm a moron.
I had two eac3to folders. One updated, and another one from which I extracted the movie, who wasn't.
Now I have only one folder, updated, and with it I extract the movie flawlessly.

Sorry I have wasted your time.
At least everything is fine now. Thanks anyway

[tebasuna51]

Sometimes I get Invalid bitstream format error with DTS to AC3 conversion and I have to use ffmpeg.

I want to supply you an example (https://ufile.io/ifxhm), so probably a fix will be out.

Your example seems corrupted.

Using Arcsoft show the same message:

The ArcSoft DTS Decoder reported an error while decoding. <ERROR>
Aborted at file position 372244480. <ERROR>

Using ffmpeg also show a error (but continue decoding):

[dca @ 000000000050fb40] Invalid bit allocation index
Error while decoding stream #0:0: Invalid data found when processing input

Don't exist a fix for that.

[pstn]

Does anyone know if it is still necessary to do eac3to in.dtshd out.dtshd -21ms after converting PCM > DTS-HD with DTS Master Suite?

[tebasuna51]

Like all encoders DTS Master Suite can add a initial delay to the encoded stream, but the output have a special global header than show some global info than include the delay used.

Whith my tool LeeAudBi4 (https://www.sendspace.com/file/vvdv2p) can read that info, for instance:

File ........: D:\tmp\DTSENC.dtshd
Size ........: 3330600 bytes

---------------------------------------------- Header Info
ChunkID .....: DTSHDHDR (Length: 16)
SubchunkID ..: CORESSMD (Length: 12)
Core SampleR : 48000
Core Bitrate : 1509
Core ChMask : 2059 (L,R,C,LFE,Lss,Rss)
Core FrameLen: 2012
SubchunkID ..: EXTSS_MD (Length: 8)
Ext. Avg. Bit: 388
SubchunkID ..: AUPR-HDR (Length: 24)
Samplerate...: 48000
Num. frames..: 1315
Samples frame: 512
Num samp. ori: 671919
Channel Mask.: 2123 (L,R,C,LFE,Lss,Rss,Lsr,Rsr)
Samples Delay: 1024 [1]
SubchunkID ..: STRMDATA (Length: 3324188)
Offset data .: 140 (Extrachunks at end of file: 6272 bytes) [2]
------------------------------------------ End Header Info
Delay .......: 21.333 ms., (2 frames) [1]
Duration ....: 14.027 sec., (0h. 0m. 14.027s.)
Avg. Bitrate : 1896 Kb/s.
------------------------------------------------- End Info

eac3to remove the global header DTSHDHDR and also the final Extrachunks [2], but don't apply the delay [1].
Then yes you can remove it with the parameter -21ms

[pstn]

Like all encoders DTS Master Suite can add a initial delay to the encoded stream, but the output have a special global header than show some global info than include the delay used.

Whith my tool LeeAudBi4 (https://www.sendspace.com/file/vvdv2p) can read that info, for instance:

eac3to remove the global header DTSHDHDR and also the final Extrachunks [2], but don't apply the delay [1].
Then yes you can remove it with the parameter -21ms

That's great thanks,

I just tried it on a DTS HD file made with DTS Master Suite,

this is what I got :

File ........: C:\Users\pstn\Desktop\track2_eng.dts
Size ........: 691429400 bytes

----------------------------------------- First Frame Info
CRC present .................: 0 (Not)
Number of PCM Sample Blocks .: 15 ( 512 samples/frame)
Primary Frame Byte Size .....: 2011 ( 2012 bytes/frame)
Audio Channel Arrangement ...: 9 (5 C + L + R + SL + SR)
Core Audio Samp. Frequency ..: 13 (48 kHz)
Transmission Bit Rate .......: 24 (1536 Kb/s)
Embedded Down Mix Enabled ...: 0 (Not)
Embedded Dynamic Range Flag .: 0 (Not)
Embedded Time Stamp Flag ....: 0 (Not)
Auxiliary Data Flag .........: 0 (Not)
Mastered in HDCD format .....: 0 (Not)
Extension Audio Descr. Flag .: 0 (Channel Extension XCh)
Extended Coding Flag ........: 0 (Not)
Audio Sync Word Insert. Flag : 1 (Sub-sub-frame)
Low Frequency Effects Flag ..: 2 (Present, interpolation factor 64)
Predictor History Flag Switch: 1 (Yes)
Multirate Interpolator Switch: 0 (Non-perfect Reconstruction)
Encoder Software Revision ...: 7 (Current)
Copy History ................: 1 (Definition deliberately omitted)
Source PCM Resolution .......: 0 (16 bits)
Front Sum/Difference Flag ...: 0 (Not)
Surrounds Sum/Difference Flag: 0 (Not)
Dialog Normalization Param. .: - 0 dB
--------------------------------------------- Revised Info
Final bytes after core: 716
Total Frames ......: 236770
Duration ..........: 2525.547 seconds. ( 0 h. 42 m. 5.547 s.)
Master A. min./max.: 68 / 2620 (HD-MA subframe bytes)
------------------------------------------------- End Info

For some reason it doesn't show me the delay, but it was made with DTS Master Suite, so it should have one I think right? I used "extended info" in your app.

[tebasuna51]

I just tried it on a DTS HD file made with DTS Master Suite:

File ........: C:\Users\pstn\Desktop\track2_eng.dts

You need use a .dtshd just encoded with DTS Master Suite, after mixed in a container the global header DTSHDHDR disapear.

EDIT:
MkvMerge and tsMuxer remove the global header (and end Extrachunks) of a .dtshd just created with DTS Master Suite, but don't apply the delay.
You can't know if you need apply the -21ms or not.

[pstn]

You need use a .dtshd just encoded with DTS Master Suite, after mixed in a container the global header DTSHDHDR disapear.

EDIT:
MkvMerge and tsMuxer remove the global header (and end Extrachunks) of a .dtshd just created with DTS Master Suite, but don't apply the delay.
You can't know if you need apply the -21ms or not.

I see, thank you

[tormento]

Your example seems corrupted.
As FFMPEG seems more resilient, a flag to ignore corrupted frames would be ok.

[hubblec4]

Hi madshi

eac3to don't find the correct mpls for the Bluray Spider Man Homecoming (https://forum.videohelp.com/attachments/43873-1512076591/Spider%20Man%20-%20Homecomig%20BD.7z)


mpls 00420 with 16 chapters is missing.

eac3to v3.33
command line: "D:\eac3to.exe" "K:\" -log="E:\Spider Man - Homecoming\title.txt"
------------------------------------------------------------------------------
1) 00421.mpls, 00001.m2ts, 2:13:28
- Chapters, 83 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- DTS Master Audio, English, multi-channel, 48kHz
- DTS Master Audio, German, multi-channel, 48kHz
- AC3, Turkish, multi-channel, 48kHz

2) 00422.mpls, 0:32:00
[334+342+343+344+345+346+347+348+349+350+351+352+353+354+355+356+357+335+338].m2ts
- Chapters, 17 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- AC3, English, stereo, 48kHz

3) 00353.mpls, 0:16:17
[342+343+344+345+346+347+348+349+350+351+352].m2ts
- Chapters, 10 chapters
- h264/AVC, 1080p24 /1.001 (16:9)
- AC3, English, stereo, 48kHz

[mtamimi]

I have a problem with a movie, the movie plays well w/o issues...However when I tried to convert the audio track I've had some issues...

eac3to v3.34
command line: eac3to.exe L:\Movie.REMUX.BluRay.AVC.DD.2.0.mkv -progressnumbers -log=log.1.txt
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 1:38:49, 24p /1.001
1: h264/AVC, English, 1080p24 /1.001 (16:9)
2: AC3, English, 2.0 channels, 192kbps, 48kHz
"Stereo"

As you see, it is AC3 2.0 and its 1:38:49...This is correct...Now, extracting...

eac3to v3.34
command line: eac3to.exe L:\Movie.REMUX.BluRay.AVC.DD.2.0.mkv 2:Audio.ac3 -progressnumbers -log=log.2.txt
------------------------------------------------------------------------------
MKV, 1 video track, 1 audio track, 1:38:49, 24p /1.001
1: h264/AVC, English, 1080p24 /1.001 (16:9)
2: AC3, English, 2.0 channels, 192kbps, 48kHz
"Stereo"
[a02] Extracting audio track number 2...
[a02] Creating file "Audio.ac3"...
Video track 1 contains 284321 frames.
eac3to processing took 3 minutes, 48 seconds.
Done.


Now, check this
eac3to v3.34
command line: eac3to.exe Audio.ac3 -progressnumbers -log=log.3.txt
------------------------------------------------------------------------------
AC3, 2.0 channels, 3:50:24, 192kbps, 48kHz

it reports the file as 3:50:24. If I converted the file to anything, I will have a 3:50:24 long audio track, and it sounds as if it was slowed down...

Can you guys investiate this?

The extracted audio track is here
https://mega.nz/#!9uYFmQJb!Mk3i3QY0GAVVoPLb1U6md3GuzSGvHEQGnNjA8GFv8Sw

[tebasuna51]

Can you guys investiate this?

The first 8 seconds are 2.0 the rest is:

AC3, 5.1 channels, 1:38:41, 448kbps, 48kHz

[XadoX]

@madshi Thx for the Update now I can use PowerShell without probs.

[mtamimi]

The first 8 seconds are 2.0 the rest is:

AC3, 5.1 channels, 1:38:41, 448kbps, 48kHz

Great, how did you figure this out? And what is the proper/precise way to trim the audio track ?

thnx :thanks: :thanks: :thanks:


*edit*
I tried to use your utility LaaAudBi4, Avira reported a threat and blocked it...
http://i67.tinypic.com/2lcsjyo.png

[LigH]

Heuristic warnings are usually false alarms, report them to your AV company.

Regarding automatic splitting of AC3 related to a changing channel layout, I am not sure what to do these days; I remember that ProjectX was able to detect channel layout changes, but only in SD TS with MPEG-2 video, it can't handle MKV. HeadAC3he was able to extract time ranges from AC3 files frame-wise, but you would have to know the exact timecodes.

[mtamimi]

Heuristic warnings are usually false alarms, report them to your AV company.

Regarding automatic splitting of AC3 related to a changing channel layout, I am not sure what to do these days; I remember that ProjectX was able to detect channel layout changes, but only in SD TS with MPEG-2 video, it can't handle MKV. HeadAC3he was able to extract time ranges from AC3 files frame-wise, but you would have to know the exact timecodes.

He was able to detect when the change started, so perhaps he uses a utility that will help in splitting the 2.0 track from the 5.1...?

[sneaker_ger]

Too lazy to test now but maybe you can use delay in this simple case. Something like:
eac3to input.ac3 temp.ac3 -8000ms
eac3to temp.ac3 output.ac3 +8000ms

Or with edit function. I'm sure tebasuna51 will post a proper solution(*) if this doesn't work. (If there is sound then maybe recode the first 8 seconds to 5.1)


(* if you want to regard this as a problem to begin with)

[tebasuna51]

He was able to detect when the change started, so perhaps he uses a utility that will help in splitting the 2.0 track from the 5.1...?
Try SplitAc3 from https://forum.doom9.org/showthread.php?p=1447695#post1447695

EDIT: The sneaker_ger solution work fine because the first 8 sec are silence.

[mtamimi]

Try SplitAc3 from https://forum.doom9.org/showthread.php?p=1447695#post1447695

EDIT: The sneaker_ger solution work fine because the first 8 sec are silence.

:thanks: a lot :)

*edit*
Anyone aware of a utility that helps me to trim the 1st 8000ms of the h264 video. If eac3to is capable of doing that, would someone show me an example?!

[sneaker_ger]

mkvtoolnix can trim video but it is problematic because of the keyframe limitation. Might need better tool like SolveigMM Video Splitter, VideoReDo or similar. Easiest to just leave the 8s.

[LigH]

Well, the optimal tool to split audio streams is one that doesn't care about video frames because it handles the audio only.

[sneaker_ger]

He wants to cut the 8s away altogether (video and audio).

[mtamimi]

He wants to cut the 8s away altogether (video and audio).

I did add +8000ms to the trimmed 5.1 auido track, and remuxed with the original untouched video and everything is in sync w/o issues. This is also good since I don't need to modify the chapters also...

I am only curious if I decided to trim the video by -8000ms instead of adding +8000ms of silence to the audio track...how can I do that easily?

:thanks:

[LigH]

But Dolby Digital AC-3 has 32 ms frames (at 48 kHz, with 1536 samples). Fortunately, 8000 ms are an integer multiple of its frame duration. But what if not? Will the duration be rounded up or down to the next audio frame border? It may be (generally) more reliable to split audio streams at the frame boundary between different channel layouts, no matter which duration they have written in seconds and milliseconds.

[sneaker_ger]

Depends on the software. With mkvtoolnix you can just do "--split parts:00:08-" and it will cut all tracks accordingly and also set delays in container to make precision <=1 ms if cuts aren't exactly on frame boundaries (which - like you said - seldomly align). It's not really a problem anyone should lose sleep about.

[tebasuna51]

Cut all tracks of a container without recode (VideoReDo or similar can recode only boundaries) at a exact point need:

- The time point is a exact multiple of audio framelength. Here, with AC3 48 Khz, 8000 ms are 250 x 32 ms, then cut the first 250 frames.

- The time point is a exact multiple of video framelength. If video have 25 fps you can cut the first 8000 / 25 = 320 frames, but you need than the 321ª frame was a Keyframe.
With other fps or without a Keyframe at exact point you can't cut exactly without recode.

[LigH]

But the problem of mtamimi here is not cutting at an arbitraty timecode. It is cutting at the audio frame where the channel layout changes. And it is knowing the position of that audio frame, not in milliseconds, but in audio frame number. And you already suggested SplitAc3 (https://forum.doom9.org/showthread.php?p=1447695#post1447695) for this job. All is fine. :)

[mtamimi]

Cut all tracks of a container without recode (VideoReDo or similar can recode only boundaries) at a exact point need:

- The time point is a exact multiple of audio framelength. Here, with AC3 48 Khz, 8000 ms are 250 x 32 ms, then cut the first 250 frames.

- The time point is a exact multiple of video framelength. If video have 25 fps you can cut the first 8000 / 25 = 320 frames, but you need than the 321ª frame was a Keyframe.
With other fps or without a Keyframe at exact point you can't cut exactly without recode.

"VideoReDo or similar can recode only boundaries" << Does this make a difference?

"you can't cut exactly without recode." :(

[tebasuna51]

"VideoReDo or similar can recode only boundaries" << Does this make a difference?

Recode always lose quality, but it's the best choice if you need that cut.

If you need more info please open a thread in video subforum, is off topic here.

[mtamimi]

Recode always lose quality, but it's the best choice if you need that cut.

Does not worth it. I prefer the 8s silence... :thanks:

[ashlar42]

Is there any up to date GUI available?

[pstn]

Does anyone know if there are any risks or downside to demuxing from a share (LAN) or a mounted ISO?

[LigH]

It's not much different to copying files. Mostly sequential reading from one file, and mostly sequential writing into a few other files. I would try to avoid both reading and writing over LAN due to low speed.

[Megalith]

Can someone clarify whether or not "Removing AC3 dialog normalization" affects TrueHD tracks? Or does it only affect the AC3 "core" track?

[Ripman]

I was trying to convert a .ts file to mkv. I got a message about the audio track not being "clean". See below. What does that mean? Thanks and happy new year.


eac3to v3.34
command line: eac3to war_requiem.ts 1: v.h264 2: a.flac
------------------------------------------------------------------------------
TS, 1 video track, 1 audio track, 1:30:00, 50p
1: h264/AVC, 720p50 (16:9)
2: AC3, German, 2.0 channels, 448kbps, 48kHz, dialnorm: -23dB, 9ms
[v01] Extracting video track number 1...
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[a02] Decoding with libav/ffmpeg...
[a02] Applying RAW/PCM delay...
[a02] Reducing depth from 64 to 24 bits...
[a02] Encoding FLAC with libFlac...
[v01] Creating file "v.h264"...
[a02] This track is not clean. <WARNING>
[a02] Creating file "a.flac"...
[a02] Audio has a gap of 32ms at playtime 0:00:02. <WARNING>
[a02] Starting 2nd pass...
[a02] Extracting audio track number 2...
[a02] Removing AC3 dialog normalization...
[a02] Decoding with libav/ffmpeg...
[a02] Applying RAW/PCM delay...
[a02] Reducing depth from 64 to 24 bits...
[a02] Encoding FLAC with libFlac...
[a02] Realizing RAW/PCM gaps...
[a02] Creating file "a.flac"...
Video track 1 contains 270021 frames.
eac3to processing took 1 hour, 18 minutes.
Done.


[edit] the output flac from the command above was 270min - was supposed to be 90 min. Must be a bad audio stream in that .Ts file. I’ll post the mediainfo.

[edit2]mediainfo -f

General
Count : 327
Count of stream of this kind : 1
Kind of stream : General
Kind of stream : General
Stream identifier : 0
ID : 1
ID : 1 (0x1)
Count of video streams : 1
Count of audio streams : 1
Video_Format_List : AVC
Video_Format_WithHint_List : AVC
Codecs Video : AVC
Audio_Format_List : AC-3
Audio_Format_WithHint_List : AC-3
Audio codecs : AC3
Audio_Language_List : German
Complete name : C:\0\War_Requiem.ts
Folder name : C:\0
File name : War_Requiem
File extension : ts
Format : MPEG-TS
Format : MPEG-TS
Format/Extensions usually used : ts m2t m2s m4t m4s tmf ts tp trp ty
Commercial name : MPEG-TS
Internet media type : video/MP2T
Codec : MPEG-TS
Codec : MPEG-TS
Codec/Extensions usually used : ts m2t m2s m4t m4s tmf ts tp trp ty
File size : 6321633292
File size : 5.89 GiB
File size : 6 GiB
File size : 5.9 GiB
File size : 5.89 GiB
File size : 5.887 GiB
Duration : 5400380.000000
Duration : 1 h 30 min
Duration : 1 h 30 min 0 s 380 ms
Duration : 1 h 30 min
Duration : 01:30:00.380
Duration : 01:30:00:23
Duration : 01:30:00.380 (01:30:00:23)
Overall bit rate mode : VBR
Overall bit rate mode : Variable
Overall bit rate : 9364703
Overall bit rate : 9 365 kb/s
Maximum Overall bit rate : 35500000
Maximum Overall bit rate : 35.5 Mb/s
Frame rate : 50.000
Frame rate : 50.000 FPS
Frame count : 270023
Stream size : 315367443
Stream size : 301 MiB (5%)
Stream size : 301 MiB
Stream size : 301 MiB
Stream size : 301 MiB
Stream size : 300.8 MiB
Stream size : 301 MiB (5%)
Proportion of this stream : 0.04989
File creation date : UTC 2017-12-31 22:57:08.222
File creation date (local) : 2017-12-31 17:57:08.222
File last modification date : UTC 2015-10-16 14:42:01.150
File last modification date (local) : 2015-10-16 09:42:01.150
OverallBitRate_Precision_Min : 9364702
OverallBitRate_Precision_Max : 9364704

Video
Count : 338
Count of stream of this kind : 1
Kind of stream : Video
Kind of stream : Video
Stream identifier : 0
StreamOrder : 0-0
ID : 4113
ID : 4113 (0x1011)
Menu ID : 1
Menu ID : 1 (0x1)
Format : AVC
Format/Info : Advanced Video Codec
Format/Url : http://developers.videolan.org/x264.html
Commercial name : AVC
Format profile : High@L4
Format settings : CABAC / 4 Ref Frames
Format settings, CABAC : Yes
Format settings, CABAC : Yes
Format settings, ReFrames : 4
Format settings, ReFrames : 4 frames
Format settings, GOP : M=4, N=32
Internet media type : video/H264
Codec ID : 27
Codec : AVC
Codec : AVC
Codec/Family : AVC
Codec/Info : Advanced Video Codec
Codec/Url : http://developers.videolan.org/x264.html
Codec profile : High@L4
Codec settings : CABAC / 4 Ref Frames
Codec settings, CABAC : Yes
Codec_Settings_RefFrames : 4
Duration : 5400460
Duration : 1 h 30 min
Duration : 1 h 30 min 0 s 460 ms
Duration : 1 h 30 min
Duration : 01:30:00.460
Duration : 01:30:00:23
Duration : 01:30:00.460 (01:30:00:23)
Bit rate mode : VBR
Bit rate mode : Variable
Bit rate : 8449420
Bit rate : 8 449 kb/s
Maximum bit rate : 40000000
Maximum bit rate : 40.0 Mb/s
Width : 1280
Width : 1 280 pixels
Height : 720
Height : 720 pixels
Sampled_Width : 1280
Sampled_Height : 720
Pixel aspect ratio : 1.000
Display aspect ratio : 1.778
Display aspect ratio : 16:9
Frame rate : 50.000
Frame rate : 50.000 FPS
Frame count : 270023
Resolution : 8
Resolution : 8 bits
Colorimetry : 4:2:0
Color space : YUV
Chroma subsampling : 4:2:0
Chroma subsampling : 4:2:0
Bit depth : 8
Bit depth : 8 bits
Scan type : Progressive
Scan type : Progressive
Interlacement : PPF
Interlacement : Progressive
Bits/(Pixel*Frame) : 0.183
Delay : 600000.000
Delay : 10 min 0 s
Delay : 10 min 0 s 0 ms
Delay : 10 min 0 s
Delay : 00:10:00.000
Delay, origin : Container
Delay, origin : Container
Stream size : 5703844345
Stream size : 5.31 GiB (90%)
Stream size : 5 GiB
Stream size : 5.3 GiB
Stream size : 5.31 GiB
Stream size : 5.312 GiB
Stream size : 5.31 GiB (90%)
Proportion of this stream : 0.90227
Buffer size : 30000000

Audio
Count : 306
Count of stream of this kind : 1
Kind of stream : Audio
Kind of stream : Audio
Stream identifier : 0
StreamOrder : 0-1
ID : 4352
ID : 4352 (0x1100)
Menu ID : 1
Menu ID : 1 (0x1)
Format : AC-3
Format/Info : Audio Coding 3
Commercial name : AC-3
Format settings, Endianness : Big
Codec ID : 129
Codec : AC3
Codec : AC3
Duration : 5400384
Duration : 1 h 30 min
Duration : 1 h 30 min 0 s 384 ms
Duration : 1 h 30 min
Duration : 01:30:00.384
Duration : 01:30:43:29
Duration : 01:30:00.384 (01:30:43:29)
Bit rate mode : CBR
Bit rate mode : Constant
Bit rate : 448000
Bit rate : 448 kb/s
Channel(s) : 2
Channel(s) : 2 channels
Channel positions : Front: L R
Channel positions : 2/0/0
ChannelLayout : L R
Samples per frame : 1536
Sampling rate : 48000
Sampling rate : 48.0 kHz
Samples count : 259218432
Frame rate : 31.250
Frame rate : 31.250 FPS (1536 spf)
Frame count : 168762
Resolution : 16
Resolution : 16 bits
Bit depth : 16
Bit depth : 16 bits
Compression mode : Lossy
Compression mode : Lossy
Delay : 600009.000
Delay : 10 min 0 s
Delay : 10 min 0 s 9 ms
Delay : 10 min 0 s
Delay : 00:10:00.009
Delay, origin : Container
Delay, origin : Container
Delay relative to video : 9
Delay relative to video : 9 ms
Delay relative to video : 9 ms
Delay relative to video : 9 ms
Delay relative to video : 00:00:00.009
Video0 delay : 9
Video0 delay : 9 ms
Video0 delay : 9 ms
Video0 delay : 9 ms
Video0 delay : 00:00:00.009
Stream size : 302421504
Stream size : 288 MiB (5%)
Stream size : 288 MiB
Stream size : 288 MiB
Stream size : 288 MiB
Stream size : 288.4 MiB
Stream size : 288 MiB (5%)
Proportion of this stream : 0.04784
Language : de
Language : German
Language : German
Language : de
Language : deu
Language : de
Service kind : CM
Service kind : Complete Main
bsid : 6
dialnorm : -23
dialnorm : -23 dB
compr : -0.28
compr : -0.28 dB
dsurmod : 1
dsurmod : Not Dolby Surround encoded
acmod : 2
lfeon : 0
dialnorm_Average : -23
dialnorm_Average : -23 dB
dialnorm_Minimum : -27
dialnorm_Minimum : -27 dB
dialnorm_Maximum : -23
dialnorm_Maximum : -23 dB
dialnorm_Count : 1885
compr_Average : -0.13
compr_Average : -0.13 dB
compr_Minimum : -0.28
compr_Minimum : -0.28 dB
compr_Maximum : 2.36
compr_Maximum : 2.36 dB
compr_Count : 1885
dynrng_Average : 0.11
dynrng_Average : 0.11 dB
dynrng_Minimum : 0.00
dynrng_Minimum : 0.00 dB
dynrng_Maximum : 2.15
dynrng_Maximum : 2.15 dB
dynrng_Count : 1885
format_identifier : AC-3

[tebasuna51]

...the output flac from the command above was 270min - was supposed to be 90 min.

Your AC3 seems have some frames 2.0, but most of them are 5.1.

Extract and split the AC3 track.
http://forum.doom9.org/showthread.php?p=1447695#post1447695

[Ripman]

Your AC3 seems have some frames 2.0, but most of them are 5.1.

Extract and split the AC3 track.
http://forum.doom9.org/showthread.php?p=1447695#post1447695

Thanks for that tip. Which line in the mediainfo report caught your attention? I tried the SplitAc3 program. Here is the log:


FileSize : 302421504 bytes
---------- First valid Header
Time eq. : 5400384 ms.
SamplCod : 0 (0:48, 1:44.1, 2:32 KHz.)
BitRate : 448 Kb/s
ChanMode : 2 (1:1/0, 2:2/0, 3:3/0, 4:2/1, 5:3/1, 6:2/2, 7:3/2)
FrameSize: 1792 bytes
---------- Process ( 25.000000 fps is used for Trim)
Time: 0 ms. Written: 69 frames 2.0 ( 2208 ms.)
Time: 2208 ms. Written: 2 frames 5.1 ( 64 ms.) Trim(55, 56)
Time: 2272 ms. Skipped: 1792 bytes (Maybe: 32 ms.)
Time: 2304 ms. Written: 168665 frames 5.1 ( 5397280 ms.) Trim(58, 134989)
Time: 5399584 ms. Written: 6 frames 2.0 ( 192 ms.)
Time: 5399776 ms. Written: 9 frames 5.1 ( 288 ms.) Trim(134994, 135000)
Time: 5400064 ms. Written: 1 frames 2.0 ( 32 ms.)
Time: 5400096 ms. Written: 3 frames 5.1 ( 96 ms.) Trim(135002, 135003)
Time: 5400192 ms. Skipped: 1792 bytes (Maybe: 32 ms.)
Time: 5400224 ms. Written: 5 frames 2.0 ( 160 ms.)
---------- End of File
Total time: 5400384 ms. at EOF
T. written: 168679 frames 5.1.
T. written: 81 frames 2.0.


[edit] looking at the splitac3 log, it seems like my tv signal got a bit corrupted — a little at the start, and a little at the end. It seems like splitac3 put 2ch samples in one output file, and 5.1ch samples in another file.

I could go back to the source .ts file, trim 2.304 secs from the start, preserve the next 5397280 ms (01:29:57.28), discard everything after that, and then extract the audio. Or, I could "zero-out" the first 2 frames (64ms) from the SplitAc3 5.1ch output, and everything after (64ms + 5397280 ms). Does this sound about right? How do I account for delays in the original audio stream? Thanks again.

[tebasuna51]

Which line in the mediainfo report caught your attention?
Nothing in MediaInfo.
Is a typical problem, the output length is 3 times the expected because 5.1 have 3 times samples than 2.0

I could go back to the source .ts file, trim 2.304 secs from the start, preserve the next 5397280 ms (01:29:57.28), discard everything after that, and then extract the audio. Or, I could "zero-out" the first 2 frames (64ms) from the SplitAc3 5.1ch output, and everything after (64ms + 5397280 ms). Does this sound about right? How do I account for delays in the original audio stream? Thanks again.

If you don't need the video I think is better cut the ts file, maybe commercials.

But if the video is OK you can use the 5.1 extracted by SplitAc3 with:

eac3to split51.ac3 one.ac3 -64ms
(the first 2 frames)
eac3to one.ac3 two.ac3 +2304ms
(audio/video delay)

And, maybe, you can delete the last 12 frames with DelayCut, only if sound bad.